From a5ce88909d3007caa7b65996a8f6784350beb2a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Jul 2007 15:42:26 +0200 Subject: [ALSA] Clean up with common snd_ctl_boolean_*_info callbacks Clean up codes using the new common snd_ctl_boolean_*_info() callbacks. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4d7f8d11ad75..fafadf9fab8e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -350,15 +350,7 @@ static struct hda_codec_ops ad198x_patch_ops = { * EAPD control * the private value = nid | (invert << 8) */ -static int ad198x_eapd_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} +#define ad198x_eapd_info snd_ctl_boolean_mono_info static int ad198x_eapd_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit From bddcf5411ffd17bfb86c2baed4a1b859c7071c98 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Jul 2007 18:04:05 +0200 Subject: [ALSA] hda-codec - Fix AD1988 SPDIF output The SPDIF output on AD1988 had some problems due to the wrongly routed analog loopback to SPDIF. This patch fixes the implementation of 'IEC958 Playback Source' mixer to handle the amp bits of mixer widget 0x1d correctly. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 45 ++++++++++++++++++++++++++++++-------------- 1 file changed, 31 insertions(+), 14 deletions(-) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fafadf9fab8e..488724f2e304 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1889,16 +1889,19 @@ static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int sel; - sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0); - if (sel > 0) { + sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT); + if (!(sel & 0x80)) + ucontrol->value.enumerated.item[0] = 0; + else { sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0); if (sel < 3) sel++; else sel = 0; + ucontrol->value.enumerated.item[0] = sel; } - ucontrol->value.enumerated.item[0] = sel; return 0; } @@ -1910,17 +1913,32 @@ static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol, int change; val = ucontrol->value.enumerated.item[0]; - sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0); if (!val) { - change = sel != 0; - if (change || codec->in_resume) - snd_hda_codec_write(codec, 0x02, 0, - AC_VERB_SET_CONNECT_SEL, 0); + sel = snd_hda_codec_read(codec, 0x1d, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT); + change = sel & 0x80; + if (change || codec->in_resume) { + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(1)); + } } else { - change = sel == 0; - if (change || codec->in_resume) - snd_hda_codec_write(codec, 0x02, 0, - AC_VERB_SET_CONNECT_SEL, 1); + sel = snd_hda_codec_read(codec, 0x1d, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT | 0x01); + change = sel & 0x80; + if (change || codec->in_resume) { + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(0)); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + } sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0) + 1; change |= sel != val; @@ -2039,10 +2057,9 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* SPDIF out pin */ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x17}, /* 0dB */ { } }; -- cgit From 532d5381793f3c824f8ff68d7067fab8c76bb811 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Jul 2007 19:02:40 +0200 Subject: [ALSA] hda-codec - Add a generic bind-control helper Added callbacks for a generic bind-control of mixer elements. This can be used for creating a mixer element controlling multiple widgets at the same time. Two macros, HDA_BIND_VOL() and HDA_BIND_SW(), are introduced for creating bind-volume and bind-switch, respectively. It taks the mixer element name and struct hda_bind_ctls pointer, which contains the real control callbacks in ops field and long array for private_value of each bound widget. All widgets have to be the same type (i.e. the same amp capability). Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 177 ++++++++++--------------------------------- 1 file changed, 41 insertions(+), 136 deletions(-) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 488724f2e304..cc2e944cc59f 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -422,94 +422,36 @@ static struct hda_input_mux ad1986a_capture_source = { }, }; -/* - * PCM control - * - * bind volumes/mutes of 3 DACs as a single PCM control for simplicity - */ - -#define ad1986a_pcm_amp_vol_info snd_hda_mixer_amp_volume_info - -static int ad1986a_pcm_amp_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *ad = codec->spec; - - mutex_lock(&ad->amp_mutex); - snd_hda_mixer_amp_volume_get(kcontrol, ucontrol); - mutex_unlock(&ad->amp_mutex); - return 0; -} -static int ad1986a_pcm_amp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *ad = codec->spec; - int i, change = 0; - - mutex_lock(&ad->amp_mutex); - for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) { - kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT); - change |= snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); - } - kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT); - mutex_unlock(&ad->amp_mutex); - return change; -} - -#define ad1986a_pcm_amp_sw_info snd_hda_mixer_amp_switch_info - -static int ad1986a_pcm_amp_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *ad = codec->spec; - - mutex_lock(&ad->amp_mutex); - snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); - mutex_unlock(&ad->amp_mutex); - return 0; -} - -static int ad1986a_pcm_amp_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *ad = codec->spec; - int i, change = 0; +static struct hda_bind_ctls ad1986a_bind_pcm_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT), + 0 + }, +}; - mutex_lock(&ad->amp_mutex); - for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) { - kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT); - change |= snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - } - kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT); - mutex_unlock(&ad->amp_mutex); - return change; -} +static struct hda_bind_ctls ad1986a_bind_pcm_sw = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT), + 0 + }, +}; /* * mixers */ static struct snd_kcontrol_new ad1986a_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Volume", - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | - SNDRV_CTL_ELEM_ACCESS_TLV_READ | - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, - .info = ad1986a_pcm_amp_vol_info, - .get = ad1986a_pcm_amp_vol_get, - .put = ad1986a_pcm_amp_vol_put, - .tlv = { .c = snd_hda_mixer_amp_tlv }, - .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT) - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PCM Playback Switch", - .info = ad1986a_pcm_amp_sw_info, - .get = ad1986a_pcm_amp_sw_get, - .put = ad1986a_pcm_amp_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT) - }, + /* + * bind volumes/mutes of 3 DACs as a single PCM control for simplicity + */ + HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol), + HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw), HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT), @@ -596,41 +538,23 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { /* laptop-eapd model - 2ch only */ /* master controls both pins 0x1a and 0x1b */ -static int ad1986a_laptop_master_vol_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; - - change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, - 0x7f, valp[0] & 0x7f); - change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, - 0x7f, valp[1] & 0x7f); - snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, - 0x7f, valp[0] & 0x7f); - snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, - 0x7f, valp[1] & 0x7f); - return change; -} - -static int ad1986a_laptop_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; +static struct hda_bind_ctls ad1986a_laptop_master_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + 0, + }, +}; - change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, - 0x80, valp[0] ? 0 : 0x80); - change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, - 0x80, valp[1] ? 0 : 0x80); - snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, - 0x80, valp[0] ? 0 : 0x80); - snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, - 0x80, valp[1] ? 0 : 0x80); - return change; -} +static struct hda_bind_ctls ad1986a_laptop_master_sw = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + 0, + }, +}; static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { .num_items = 3, @@ -642,23 +566,8 @@ static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { }; static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Volume", - .info = snd_hda_mixer_amp_volume_info, - .get = snd_hda_mixer_amp_volume_get, - .put = ad1986a_laptop_master_vol_put, - .tlv = { .c = snd_hda_mixer_amp_tlv }, - .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1986a_laptop_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - }, + HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), + HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT), @@ -856,7 +765,6 @@ static int patch_ad1986a(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 6; @@ -1064,7 +972,6 @@ static int patch_ad1983(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -1466,7 +1373,6 @@ static int patch_ad1981(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -2672,7 +2578,6 @@ static int patch_ad1988(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - mutex_init(&spec->amp_mutex); codec->spec = spec; if (is_rev2(codec)) -- cgit From 82beb8fd365afe3891b277c46425083f13e23c56 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Aug 2007 17:09:26 +0200 Subject: [ALSA] hda-codec - optimize resume using caches So far, the driver looked the table of snd_kcontrol_new used for creating mixer elements and forces to call each of its put callbacks in PM resume code. This is too ugly and hackish. Now, the resume is simplified using the codec amp and command register caches. The driver simply restores the values that have been written in the cache table. With this simplification, most codec support codes don't require any special resume callback. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 68 ++++++++++++++++---------------------------- 1 file changed, 25 insertions(+), 43 deletions(-) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cc2e944cc59f..f20ddd85db22 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -318,31 +318,11 @@ static void ad198x_free(struct hda_codec *codec) kfree(codec->spec); } -#ifdef CONFIG_PM -static int ad198x_resume(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - int i; - - codec->patch_ops.init(codec); - for (i = 0; i < spec->num_mixers; i++) - snd_hda_resume_ctls(codec, spec->mixers[i]); - if (spec->multiout.dig_out_nid) - snd_hda_resume_spdif_out(codec); - if (spec->dig_in_nid) - snd_hda_resume_spdif_in(codec); - return 0; -} -#endif - static struct hda_codec_ops ad198x_patch_ops = { .build_controls = ad198x_build_controls, .build_pcms = ad198x_build_pcms, .init = ad198x_init, .free = ad198x_free, -#ifdef CONFIG_PM - .resume = ad198x_resume, -#endif }; @@ -376,12 +356,12 @@ static int ad198x_eapd_put(struct snd_kcontrol *kcontrol, eapd = ucontrol->value.integer.value[0]; if (invert) eapd = !eapd; - if (eapd == spec->cur_eapd && ! codec->in_resume) + if (eapd == spec->cur_eapd) return 0; spec->cur_eapd = eapd; - snd_hda_codec_write(codec, nid, - 0, AC_VERB_SET_EAPD_BTLENABLE, - eapd ? 0x02 : 0x00); + snd_hda_codec_write_cache(codec, nid, + 0, AC_VERB_SET_EAPD_BTLENABLE, + eapd ? 0x02 : 0x00); return 1; } @@ -882,8 +862,9 @@ static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ if (spec->spdif_route != ucontrol->value.enumerated.item[0]) { spec->spdif_route = ucontrol->value.enumerated.item[0]; - snd_hda_codec_write(codec, spec->multiout.dig_out_nid, 0, - AC_VERB_SET_CONNECT_SEL, spec->spdif_route); + snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0, + AC_VERB_SET_CONNECT_SEL, + spec->spdif_route); return 1; } return 0; @@ -1824,33 +1805,34 @@ static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol, AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_INPUT); change = sel & 0x80; - if (change || codec->in_resume) { - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_UNMUTE(0)); - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_MUTE(1)); + if (change) { + snd_hda_codec_write_cache(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + snd_hda_codec_write_cache(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(1)); } } else { sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_INPUT | 0x01); change = sel & 0x80; - if (change || codec->in_resume) { - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_MUTE(0)); - snd_hda_codec_write(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_UNMUTE(1)); + if (change) { + snd_hda_codec_write_cache(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(0)); + snd_hda_codec_write_cache(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); } sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0) + 1; change |= sel != val; - if (change || codec->in_resume) - snd_hda_codec_write(codec, 0x0b, 0, - AC_VERB_SET_CONNECT_SEL, val - 1); + if (change) + snd_hda_codec_write_cache(codec, 0x0b, 0, + AC_VERB_SET_CONNECT_SEL, + val - 1); } return change; } -- cgit From 47fd830acf0b6b5bc75db55d0f2cc64f59a23b5f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Aug 2007 17:11:07 +0200 Subject: [ALSA] hda-codec - add snd_hda_codec_stereo() function Added snd_hda_codec_amp_stereo() function that changes both of stereo channels with the same mask and value bits. It simplifies most of amp-handling codes. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index f20ddd85db22..febc2053f08e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1120,10 +1120,9 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 0; /* toggle HP mute appropriately */ - snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0, - 0x80, spec->cur_eapd ? 0 : 0x80); - snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0, - 0x80, spec->cur_eapd ? 0 : 0x80); + snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + spec->cur_eapd ? 0 : HDA_AMP_MUTE); return 1; } @@ -1136,13 +1135,13 @@ static int ad1981_hp_master_vol_put(struct snd_kcontrol *kcontrol, int change; change = snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, - 0x7f, valp[0] & 0x7f); + HDA_AMP_VOLMASK, valp[0]); change |= snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, - 0x7f, valp[1] & 0x7f); + HDA_AMP_VOLMASK, valp[1]); snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0, - 0x7f, valp[0] & 0x7f); + HDA_AMP_VOLMASK, valp[0]); snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0, - 0x7f, valp[1] & 0x7f); + HDA_AMP_VOLMASK, valp[1]); return change; } @@ -1153,10 +1152,8 @@ static void ad1981_hp_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x06, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); - snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, - 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } /* toggle input of built-in and mic jack appropriately */ -- cgit From cca3b3718ca96dca51daf1129ac03003bcede751 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Aug 2007 17:12:15 +0200 Subject: [ALSA] hda-codec - Clean up bind-controls We have already a generic bind-control helper, so let's clean up the codes using it. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 34 +++++++++------------------------- 1 file changed, 9 insertions(+), 25 deletions(-) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index febc2053f08e..f9390a544ea4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1127,23 +1127,14 @@ static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, } /* bind volumes of both NID 0x05 and 0x06 */ -static int ad1981_hp_master_vol_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - long *valp = ucontrol->value.integer.value; - int change; - - change = snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, - HDA_AMP_VOLMASK, valp[0]); - change |= snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, - HDA_AMP_VOLMASK, valp[1]); - snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0, - HDA_AMP_VOLMASK, valp[0]); - snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0, - HDA_AMP_VOLMASK, valp[1]); - return change; -} +static struct hda_bind_ctls ad1981_hp_bind_master_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT), + 0 + }, +}; /* mute internal speaker if HP is plugged */ static void ad1981_hp_automute(struct hda_codec *codec) @@ -1204,14 +1195,7 @@ static struct hda_input_mux ad1981_hp_capture_source = { }; static struct snd_kcontrol_new ad1981_hp_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Volume", - .info = snd_hda_mixer_amp_volume_info, - .get = snd_hda_mixer_amp_volume_get, - .put = ad1981_hp_master_vol_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), - }, + HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", -- cgit From cb53c626e1145edf1d619bc4953f6293d3a77ace Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Aug 2007 17:21:45 +0200 Subject: [ALSA] hda-intel - Add POWER_SAVE option Added CONFIG_SND_HDA_POWER_SAVE kconfig. It's an experimental option to achieve an aggressive power-saving. With this option, the driver will turn on/off the power of each codec and controller chip dynamically on demand. The patch introduces a new module option 'power_save'. It specifies the second of time-out for automatic power-down. As default, it's 10 seconds. Setting 0 means to suppress the power-saving feature. The codec may have analog-input loopbacks, which are usually represented by mixer elements such as 'Mic Playback Switch' or 'CD Playback Switch'. When these are on, we cannot turn off the mixer and the codec chip has to be kept on. For bookkeeping these states, a new codec-callback is introduced. For the bus-controller side, a new callback pm_notify is introduced, which can be used to turn on/off the contoller appropriately. Note that this power-saving might cause slight click-noise at power-on/off. Also, it might take some time to wake up the codec, and might even drop some tones at the very beginning. This seems to be the side-effect of turning off the controller chip. This turn-off of the controller can be disabled by undefining HDA_POWER_SAVE_RESET_CONTOLLER in hda_intel.c. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 91 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 91 insertions(+) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index f9390a544ea4..53cfa0da4964 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -73,6 +73,10 @@ struct ad198x_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[4]; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_loopback_check loopback; +#endif }; /* @@ -144,6 +148,14 @@ static int ad198x_build_controls(struct hda_codec *codec) return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + struct ad198x_spec *spec = codec->spec; + return snd_hda_check_amp_list_power(codec, &spec->loopback, nid); +} +#endif + /* * Analog playback callbacks */ @@ -323,6 +335,9 @@ static struct hda_codec_ops ad198x_patch_ops = { .build_pcms = ad198x_build_pcms, .init = ad198x_init, .free = ad198x_free, +#ifdef CONFIG_SND_HDA_POWER_SAVE + .check_power_status = ad198x_check_power_status, +#endif }; @@ -736,6 +751,17 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { {} }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list ad1986a_loopbacks[] = { + { 0x13, HDA_OUTPUT, 0 }, /* Mic */ + { 0x14, HDA_OUTPUT, 0 }, /* Phone */ + { 0x15, HDA_OUTPUT, 0 }, /* CD */ + { 0x16, HDA_OUTPUT, 0 }, /* Aux */ + { 0x17, HDA_OUTPUT, 0 }, /* Line */ + { } /* end */ +}; +#endif + static int patch_ad1986a(struct hda_codec *codec) { struct ad198x_spec *spec; @@ -759,6 +785,9 @@ static int patch_ad1986a(struct hda_codec *codec) spec->mixers[0] = ad1986a_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = ad1986a_init_verbs; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = ad1986a_loopbacks; +#endif codec->patch_ops = ad198x_patch_ops; @@ -944,6 +973,13 @@ static struct hda_verb ad1983_init_verbs[] = { { } /* end */ }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list ad1983_loopbacks[] = { + { 0x12, HDA_OUTPUT, 0 }, /* Mic */ + { 0x13, HDA_OUTPUT, 0 }, /* Line */ + { } /* end */ +}; +#endif static int patch_ad1983(struct hda_codec *codec) { @@ -968,6 +1004,9 @@ static int patch_ad1983(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1983_init_verbs; spec->spdif_route = 0; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = ad1983_loopbacks; +#endif codec->patch_ops = ad198x_patch_ops; @@ -1091,6 +1130,17 @@ static struct hda_verb ad1981_init_verbs[] = { { } /* end */ }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list ad1981_loopbacks[] = { + { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */ + { 0x13, HDA_OUTPUT, 0 }, /* Line */ + { 0x1b, HDA_OUTPUT, 0 }, /* Aux */ + { 0x1c, HDA_OUTPUT, 0 }, /* Mic */ + { 0x1d, HDA_OUTPUT, 0 }, /* CD */ + { } /* end */ +}; +#endif + /* * Patch for HP nx6320 * @@ -1350,6 +1400,9 @@ static int patch_ad1981(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1981_init_verbs; spec->spdif_route = 0; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = ad1981_loopbacks; +#endif codec->patch_ops = ad198x_patch_ops; @@ -2103,6 +2156,15 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) snd_hda_sequence_write(codec, ad1988_laptop_hp_off); } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list ad1988_loopbacks[] = { + { 0x20, HDA_INPUT, 0 }, /* Front Mic */ + { 0x20, HDA_INPUT, 1 }, /* Line */ + { 0x20, HDA_INPUT, 4 }, /* Mic */ + { 0x20, HDA_INPUT, 6 }, /* CD */ + { } /* end */ +}; +#endif /* * Automatic parse of I/O pins from the BIOS configuration @@ -2647,6 +2709,9 @@ static int patch_ad1988(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1988_laptop_unsol_event; break; } +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = ad1988_loopbacks; +#endif return 0; } @@ -2803,6 +2868,16 @@ static struct hda_verb ad1884_init_verbs[] = { { } /* end */ }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list ad1884_loopbacks[] = { + { 0x20, HDA_INPUT, 0 }, /* Front Mic */ + { 0x20, HDA_INPUT, 1 }, /* Mic */ + { 0x20, HDA_INPUT, 2 }, /* CD */ + { 0x20, HDA_INPUT, 4 }, /* Docking */ + { } /* end */ +}; +#endif + static int patch_ad1884(struct hda_codec *codec) { struct ad198x_spec *spec; @@ -2827,6 +2902,9 @@ static int patch_ad1884(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1884_init_verbs; spec->spdif_route = 0; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = ad1884_loopbacks; +#endif codec->patch_ops = ad198x_patch_ops; @@ -3208,6 +3286,16 @@ static struct hda_verb ad1882_init_verbs[] = { { } /* end */ }; +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list ad1882_loopbacks[] = { + { 0x20, HDA_INPUT, 0 }, /* Front Mic */ + { 0x20, HDA_INPUT, 1 }, /* Mic */ + { 0x20, HDA_INPUT, 4 }, /* Line */ + { 0x20, HDA_INPUT, 6 }, /* CD */ + { } /* end */ +}; +#endif + /* models */ enum { AD1882_3STACK, @@ -3246,6 +3334,9 @@ static int patch_ad1882(struct hda_codec *codec) spec->num_init_verbs = 1; spec->init_verbs[0] = ad1882_init_verbs; spec->spdif_route = 0; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = ad1882_loopbacks; +#endif codec->patch_ops = ad198x_patch_ops; -- cgit From 20a45e8644ef4f5e7dfd727859301c4c581e9489 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Aug 2007 22:20:45 +0200 Subject: [ALSA] hda-codec - Fix Master volume with AD1986A laptop model Use the bind-control for NID 0x1a and 0x1b as Master volume control on AD1986 model=laptop as well as model=laptop-eapd. This will fix the missing output on some devices. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 44 +++++++++++++++++++++----------------------- 1 file changed, 21 insertions(+), 23 deletions(-) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 53cfa0da4964..bc4b797aa97b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -498,13 +498,30 @@ static struct snd_kcontrol_new ad1986a_3st_mixers[] = { /* laptop model - 2ch only */ static hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC }; +/* master controls both pins 0x1a and 0x1b */ +static struct hda_bind_ctls ad1986a_laptop_master_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + 0, + }, +}; + +static struct hda_bind_ctls ad1986a_laptop_master_sw = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + 0, + }, +}; + static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Master Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), */ + HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), + HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT), @@ -532,25 +549,6 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { /* laptop-eapd model - 2ch only */ -/* master controls both pins 0x1a and 0x1b */ -static struct hda_bind_ctls ad1986a_laptop_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - 0, - }, -}; - -static struct hda_bind_ctls ad1986a_laptop_master_sw = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - 0, - }, -}; - static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { .num_items = 3, .items = { -- cgit From 8ab78c7424588c6b1600dcfd70418617a09326b8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 6 Sep 2007 14:29:53 +0200 Subject: [ALSA] hda-codec - Add laptop-automute model for AD1986A Added a new model laptop-automute for AD1986A, which has the HP jack detection and auto-muting of the speaker. Currently, it's used for Lenovo N100. Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 126 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 125 insertions(+), 1 deletion(-) (limited to 'sound/pci/hda/patch_analog.c') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index bc4b797aa97b..54cfd4526d20 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -74,6 +74,8 @@ struct ad198x_spec { struct hda_input_mux private_imux; hda_nid_t private_dac_nids[4]; + unsigned int jack_present :1; + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -588,6 +590,106 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { { } /* end */ }; +/* laptop-automute - 2ch only */ + +static void ad1986a_update_hp(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + unsigned int mute; + + if (spec->jack_present) + mute = HDA_AMP_MUTE; /* mute internal speaker */ + else + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); +} + +static void ad1986a_hp_automute(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + ad1986a_update_hp(codec); +} + +#define AD1986A_HP_EVENT 0x37 + +static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) != AD1986A_HP_EVENT) + return; + ad1986a_hp_automute(codec); +} + +static int ad1986a_hp_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1986a_hp_automute(codec); + return 0; +} + +/* bind hp and internal speaker mute (with plug check) */ +static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp[0] ? 0 : HDA_AMP_MUTE); + change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp[1] ? 0 : HDA_AMP_MUTE); + if (change) + ad1986a_update_hp(codec); + return change; +} + +static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { + HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1986a_hp_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "External Amplifier", + .info = ad198x_eapd_info, + .get = ad198x_eapd_get, + .put = ad198x_eapd_put, + .private_value = 0x1b | (1 << 8), /* port-D, inversed */ + }, + { } /* end */ +}; + /* * initialization verbs */ @@ -701,12 +803,20 @@ static struct hda_verb ad1986a_ultra_init[] = { { } /* end */ }; +/* pin sensing on HP jack */ +static struct hda_verb ad1986a_hp_init_verbs[] = { + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT}, + {} +}; + + /* models */ enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP, AD1986A_LAPTOP_EAPD, + AD1986A_LAPTOP_AUTOMUTE, AD1986A_ULTRA, AD1986A_MODELS }; @@ -716,6 +826,7 @@ static const char *ad1986a_models[AD1986A_MODELS] = { [AD1986A_3STACK] = "3stack", [AD1986A_LAPTOP] = "laptop", [AD1986A_LAPTOP_EAPD] = "laptop-eapd", + [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute", [AD1986A_ULTRA] = "ultra", }; @@ -744,7 +855,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), - SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_EAPD), + SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP), {} }; @@ -821,6 +932,19 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.dig_out_nid = 0; spec->input_mux = &ad1986a_laptop_eapd_capture_source; break; + case AD1986A_LAPTOP_AUTOMUTE: + spec->mixers[0] = ad1986a_laptop_automute_mixers; + spec->num_init_verbs = 3; + spec->init_verbs[1] = ad1986a_eapd_init_verbs; + spec->init_verbs[2] = ad1986a_hp_init_verbs; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1986a_laptop_dac_nids; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1986a_laptop_eapd_capture_source; + codec->patch_ops.unsol_event = ad1986a_hp_unsol_event; + codec->patch_ops.init = ad1986a_hp_init; + break; case AD1986A_ULTRA: spec->mixers[0] = ad1986a_laptop_eapd_mixers; spec->num_init_verbs = 2; -- cgit